This invention relates to the field of digital transmission and to a means for correcting errors in the reproduced signal caused by noise on the channel.
Many methods are known in the art for encoding analog signals before transmission in an attempt to make the reproduced signal an exact replica of the original signal. Some of these methods include sampling the analog signal and encoding the digitized signal in some manner. The encoding will frequently involve taking the differences between subsequent sample values, and even taking second differences in some cases. One example of this may be seen in U.S. Pat. No. 4,503,510 where the source signals are electro-cardiogram readings which are to be transmitted from the scene of an accident back to a hospital for interpretation by a doctor. Transmission of analog signals for this purpose having been found unsatisfactory, digitizing the signals had been tried. However, since the transmission path is often far from ideal, the noise in the channel often caused severe problems in interpreting the patient's symptoms. In this prior patent, the analog signals are digitized, then processed in a digital compression filter whose output is the difference between the actual value of a signal sample and an estimated value. These difference signals or delta's are then encoded using a Huffman code which takes advantage of the fact that the signals are differences and those differences have different probabilities of occurrence. Although this system operates with a greatly reduced data rate, it may be more strongly affected by noise in the channel than a simple PCM system would be. While errors in a small portion of some signals such as speech may not be too objectionable if the received signal is still intelligible, the recipient of an ECG, video or music signal may be disturbed by a single error in the received signal. The extent of the disruption will depend on the encoding/decoding scheme; in digitized audio signals it may range from a dull pop or thud to a loud noise to an unpleasant garbling of the music quality.
It is known, as shown in U.S. Pat. No. 4,055,832, to correct erroneous bits of received digital signals by sending an equal number of parity bits, but this system is inefficient of bandwidth or transmission rate.
As discussed in "Digital Audio Techniques" by Nakajima et al, 1985, Tab Books, Inc., there are many other ways to reduce or eliminate errors in digitally transmitted audio. These include using interpolation or substitution of a zero value (muting) or a previous good value or block of values for any value or block containing error. More elaborately, extrapolation--using information from both sides of the erroneous data--can be used to bridge over a deleted data word, such as creating a parabolic curve which matches the valid data at both ends. If the signal blocks are small enough these solutions will not always be objectionable, but it is far preferable to cancel the extraneous portion of the signal while retaining the correct signal. Incidentally, in this context, "error detection" typically means comparing an "update" value which is presumed to be correct with the decoded value. The update value may be arrived at in any suitable manner.